[Asterisk-Dev] SIP handling of Contact header with new port

[Ed Greenberg]

This is observed with 1.2.0, not CVS.

When a sip peer that is behind NAT is invited, it returns a contact header 
with a port other than 5060 in it's 180 and 200 messages. For instance:

    -- Called otherguy/04082098516
ev02a*CLI>
<-- SIP read from 1xx.yy.zz.aa:5060:
SIP/2.0 100 Trying
Call-ID: 64c034d2254d5edb7216047073e150c2 at 1aa.bbb.ccc.ddd
Via: SIP/2.0/UDP 1aa.bbb.ccc.ddd:5060;branch=z9hG4bK30bbe558;rport
From: "+13115552368" <sip:+13115552368 at 1aa.bbb.ccc.ddd>;tag=as3a9d8dca
To: <sip:18887776666 at 1xx.yy.zz.aa:5060>
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines)---
ev02a*CLI>
<-- SIP read from 1xx.yy.zz.aa:5060:
SIP/2.0 180 Ringing
Call-ID: 64c034d2254d5edb7216047073e150c2 at 1aa.bbb.ccc.ddd
Via: SIP/2.0/UDP 1aa.bbb.ccc.ddd:5060;branch=z9hG4bK30bbe558
From: "+13115552368" <sip:+13115552368 at 1aa.bbb.ccc.ddd>;tag=as3a9d8dca
To: <sip:18887776666 at 1xx.yy.zz.aa:5060>;tag=0a94b5e7
CSeq: 102 INVITE
Contact: <sip:18887776666 at 1xx.yy.zz.aa:32801>
Allow:  OPTIONS, INVITE, CANCEL, ACK, BYE, INFO
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 164

v=0
o=Sonus_UAC 1383 25405 IN IP4 1xx.yy.zz.aa
s=SIP Media Capabilities
c=IN IP4 1xx.yy.zz.aa
t=0 0
m=audio 32836 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

And then...

--- (11 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 1xx.yy.zz.aa:32836
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), 
combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
    -- SIP/otherguy-0c77 is ringing
    -- SIP/otherguy-0c77 is making progress passing it to 
SIP/64.156.10.50-097f01c0
ev02a*CLI>
<-- SIP read from 1xx.yy.zz.aa:5060:
SIP/2.0 200 OK
Call-ID: 64c034d2254d5edb7216047073e150c2 at 1aa.bbb.ccc.ddd
Via: SIP/2.0/UDP 1aa.bbb.ccc.ddd:5060;branch=z9hG4bK30bbe558
From: "+13115552368" <sip:+13115552368 at 1aa.bbb.ccc.ddd>;tag=as3a9d8dca
To: <sip:18887776666 at 1xx.yy.zz.aa:5060>;tag=0a94b5e7
CSeq: 102 INVITE
Accept:  multipart/mixed, application/sdp, application/isup, 
application/dtmf, application/dtmf-relay
Contact: <sip:18887776666 at 1xx.yy.zz.aa:32801>
Allow:  OPTIONS, INVITE, CANCEL, ACK, BYE, INFO
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 164

v=0
o=Sonus_UAC 1383 25405 IN IP4 1xx.yy.zz.aa
s=SIP Media Capabilities
c=IN IP4 1xx.yy.zz.aa
t=0 0
m=audio 32836 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=sendrecv

Now, below, I see set_destination setting the destination port to 32801, 
but the next packet still sends to port 5060.

--- (12 headers 8 lines)---
Found RTP audio format 0
Peer audio RTP is at port 1xx.yy.zz.aa:32836
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), 
combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
list_route: hop: <sip:18887776666 at 1xx.yy.zz.aa:32801>
set_destination: Parsing <sip:18887776666 at 1xx.yy.zz.aa:32801> for 
address/port to send to
set_destination: set destination to 1xx.yy.zz.aa, port 32801
Transmitting (NAT) to 1xx.yy.zz.aa:5060:
ACK sip:18887776666 at 1xx.yy.zz.aa:32801 SIP/2.0
Via: SIP/2.0/UDP 1aa.bbb.ccc.ddd:5060;branch=z9hG4bK29977c17;rport
From: "+13115552368" <sip:+13115552368 at 1aa.bbb.ccc.ddd>;tag=as3a9d8dca
To: <sip:18887776666 at 1xx.yy.zz.aa>;tag=0a94b5e7
Contact: <sip:+13115552368 at 1aa.bbb.ccc.ddd>
Call-ID: 64c034d2254d5edb7216047073e150c2 at 1aa.bbb.ccc.ddd
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Shouldn't this switch to talking to port 32801?

The sip.conf for this peer has nat=yes on it.
[otherguy]
type=friend
host=1xx.yy.zz.aa
context=otherguy
disallow=all
allow=ulaw
insecure=very
nat=yes

Have I found a bug?

</edg>